generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

ACM: Too short char vector

Browse Source 
Revision 2340 failed on Linux Release, and the problem was that we allocated a too short vector for the output file name.

BUG=r2340 failed on Linux release
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/624006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2343 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2012-06-01 14:51:28 +00:00
parent 343301ff73
commit 5b4f36db88
1 changed files with 6 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
src/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@@ -155,7 +155,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
} }
} }
char filename[128]; char filename[256];
_rtpStream = rtpStream; _rtpStream = rtpStream;
int playSampFreq; int playSampFreq;
@@ -300,17 +300,16 @@ void EncodeDecodeTest::Perform() {
codePars[1] = 0; codePars[1] = 0;
codePars[2] = 0; codePars[2] = 0;
AudioCodingModule *acmTmp = AudioCodingModule::Create(0); AudioCodingModule* acm = AudioCodingModule::Create(0);
struct CodecInst sendCodecTmp; struct CodecInst sendCodecTmp;
numCodecs = acmTmp->NumberOfCodecs(); numCodecs = acm->NumberOfCodecs();
AudioCodingModule::Destroy(acmTmp);
if (_testMode == 1) { if (_testMode == 1) {
printf("List of supported codec.\n"); printf("List of supported codec.\n");
} }
if (_testMode != 2) { if (_testMode != 2) {
for (int n = 0; n < numCodecs; n++) { for (int n = 0; n < numCodecs; n++) {
acmTmp->Codec(n, sendCodecTmp); acm->Codec(n, sendCodecTmp);
if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) { if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
numPars[n] = 0; numPars[n] = 0;
} else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) { } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
@@ -347,7 +346,6 @@ void EncodeDecodeTest::Perform() {
EncodeToFile(1, codeId, codePars, _testMode); EncodeToFile(1, codeId, codePars, _testMode);
AudioCodingModule *acm = AudioCodingModule::Create(10);
RTPFile rtpFile; RTPFile rtpFile;
std::string fileName = webrtc::test::OutputPath() + "outFile.rtp"; std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
rtpFile.Open(fileName.c_str(), "rb"); rtpFile.Open(fileName.c_str(), "rb");
@@ -359,13 +357,13 @@ void EncodeDecodeTest::Perform() {
_receiver.Run(); _receiver.Run();
_receiver.Teardown(); _receiver.Teardown();
rtpFile.Close(); rtpFile.Close();
AudioCodingModule::Destroy(acm);
if (_testMode == 1) { if (_testMode == 1) {
printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId); printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId);
} }
} }
} }
AudioCodingModule::Destroy(acm);
if (_testMode == 0) { if (_testMode == 0) {
printf("Done!\n"); printf("Done!\n");
} }
@@ -375,7 +373,7 @@ void EncodeDecodeTest::Perform() {
void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars, void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
int testMode) { int testMode) {
AudioCodingModule *acm = AudioCodingModule::Create(0); AudioCodingModule* acm = AudioCodingModule::Create(1);
RTPFile rtpFile; RTPFile rtpFile;
std::string fileName = webrtc::test::OutputPath() + "outFile.rtp"; std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
rtpFile.Open(fileName.c_str(), "wb+"); rtpFile.Open(fileName.c_str(), "wb+");